[SR-Users] Attended transfer in load-balanced multi-server setup
Pan B. Christensen
pan at ibidium.no
Wed Apr 13 13:49:13 CEST 2011
I'm having problems with attended transfer in certain scenarios. Let me first try to document the relevant parts of the setup:
There are two load-balanced Asterisk gateways connected to the old Nortel system. Currently, PSTN connections are done by Nortel. We then have two load-balanced kamailio servers. The phones are connected to these. There are several other servers not relevant to this question.
Blind transfer works in any scenario. I've been told that attended transfer works in any scenario if the original call is SIP to SIP. If the original call is from Nortel/PSTN to SIP, then attended transfer will usually not work.
A call comes in from Nortel/PSTN (user A) through Nortel1 to a SIP device (user B). User B pushes the transfer button (puts current call on hold and makes a new call) and dials a Nortel / PSTN number to user C. This call goes out through the nortel2 server. Users B and C talk a short while before user B pushes the Transfer button again to connect users A and C. In this scenario, the REFER is forwarded by kamailio to nortel1, which replies "SIP/2.0 481 Call leg/transaction does not exist." The call to join is of course on the other asterisk (nortel2).
A similar scenario is where user B transfers to another SIP user. This call will only exist in one of the kamailio servers, and Asterisk will give the same response to the REFER.
A third scenario is where both calls are handled by the same Asterisk server. This scenario works.
I'm assuming I'll have to build REFER-handling logic into kamailio, but am unsure of how to proceed. Any suggestions?
Thanks in advance.
With kind regards,
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